Hi
You can tray to making a tcpdump for detect where stay the audio packets (RTP) and/or stopping the iptables. Also you may check the out route (route -n) and get the default GW, it should be the Public GW.
At 09:17 a.m. 26/11/2007, Zaheer K. Master wrote:
Hi All, I'm running asterisk 1.4.5 (on AsteriskNOW beta 6 appliance) and using Snom 360/370 phones with direct SIP trunking from bandwidth.com. I can make outgoing calls, and the person on the receiving end can hear my voice, but I cannot hear them. I also cannot receive incoming calls to my DID number. Here is my current setup: Asterisk is running on a dell poweredge server with an ip of 192.168.1.55 I have setup a 1:1 NAT for asterisk with my public IP of 72.127.218.XXX The phones have IPs of 192.168.1.150-160 I can register my phones and they work correctly for intercom, voicemail, etc. I think the problem is that after the SIP session has initiated, the phones are giving an IP of 192.168.1.151 for the return audio, and those packets are getting dropped. I'm not sure where to go from here to get the incoming calls/audio working. Do I have to give the Asterisk box a public IP? I tried this, and when I did I was unable to get the phones to register - probably since they had private IPs. Any help or suggestions would be greatly appreciated. Thanks in advance! Regards, Zaheer _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. [EMAIL PROTECTED] http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph
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